a70695a3e1
The biggest change to NetEq is the move from a primary flag, to a Priority with two separate levels: one set by RED splitting and one set by the codec itself. This allows us to unambigously prioritize "fallback" packets from these two sources. I've chosen what I believe is the sensible ordering: packets that the codec prioritizes are chosen first, regardless of if they are secondary RED packets or not. So if we were to use Opus w/ FEC in RED, we'd only do Opus FEC decoding if there was no RED packet that could cover the time slot. With this change, PayloadSplitter now only deals with RED packets. Maybe it should be renamed RedPayloadSplitter? BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2342443005 Cr-Commit-Position: refs/heads/master@{#14347}
130 lines
4.5 KiB
C++
130 lines
4.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
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#include <assert.h>
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#include <memory>
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#include <utility>
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#include "webrtc/base/array_view.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/sanitizer.h"
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#include "webrtc/base/trace_event.h"
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#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
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namespace webrtc {
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AudioDecoder::ParseResult::ParseResult() = default;
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AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
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AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
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int priority,
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std::unique_ptr<EncodedAudioFrame> frame)
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: timestamp(timestamp), priority(priority), frame(std::move(frame)) {
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RTC_DCHECK_GE(priority, 0);
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}
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AudioDecoder::ParseResult::~ParseResult() = default;
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AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=(
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ParseResult&& b) = default;
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std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload(
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rtc::Buffer&& payload,
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uint32_t timestamp) {
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std::vector<ParseResult> results;
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std::unique_ptr<EncodedAudioFrame> frame(
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new LegacyEncodedAudioFrame(this, std::move(payload)));
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results.emplace_back(timestamp, 0, std::move(frame));
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return results;
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}
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int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
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int sample_rate_hz, size_t max_decoded_bytes,
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int16_t* decoded, SpeechType* speech_type) {
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TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
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rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
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int duration = PacketDuration(encoded, encoded_len);
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if (duration >= 0 &&
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duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
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return -1;
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}
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return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
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speech_type);
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}
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int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
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int sample_rate_hz, size_t max_decoded_bytes,
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int16_t* decoded, SpeechType* speech_type) {
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TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
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rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
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int duration = PacketDurationRedundant(encoded, encoded_len);
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if (duration >= 0 &&
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duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
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return -1;
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}
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return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
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speech_type);
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}
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int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
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size_t encoded_len,
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int sample_rate_hz, int16_t* decoded,
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SpeechType* speech_type) {
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return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
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speech_type);
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}
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bool AudioDecoder::HasDecodePlc() const { return false; }
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size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
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return 0;
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}
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int AudioDecoder::IncomingPacket(const uint8_t* payload,
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size_t payload_len,
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uint16_t rtp_sequence_number,
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uint32_t rtp_timestamp,
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uint32_t arrival_timestamp) {
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return 0;
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}
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int AudioDecoder::ErrorCode() { return 0; }
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int AudioDecoder::PacketDuration(const uint8_t* encoded,
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size_t encoded_len) const {
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return kNotImplemented;
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}
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int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
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size_t encoded_len) const {
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return kNotImplemented;
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}
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bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
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size_t encoded_len) const {
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return false;
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}
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AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
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switch (type) {
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case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech.
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case 1:
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return kSpeech;
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case 2:
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return kComfortNoise;
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default:
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assert(false);
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return kSpeech;
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}
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}
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} // namespace webrtc
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